Re: [SR-Users] Websocket In-dialoge SIP routing failed post network loss due to aliases

2021-06-30 Thread Juha Heinanen
Shahid Hussain writes: > Following are the REGISTER and response messages. Is it possible to > confirm the JSSIP client has full implementation of SIP outbound? Looks like it if you define two sockets. -- Juha __ Kamailio - Users Mailing

Re: [SR-Users] Websocket In-dialoge SIP routing failed post network loss due to aliases

2021-06-30 Thread Juha Heinanen
Due to other tests, I had missed baresip account's ;outbound paramater. Once I added it, also reg-id was added. -- Juha WSS 192.168.43.160:50442 -> 192.168.43.160:5061 REGISTER sip:test.tutpro.com SIP/2.0 Via: SIP/2.0/WSS 127.0.0.1:9;branch=z9hG4bK5a4ad01f9164d358;rport Contact:

Re: [SR-Users] Websocket In-dialoge SIP routing failed post network loss due to aliases

2021-06-30 Thread Juha Heinanen
Olle E. Johansson writes: > Full support for SIP outbound (using REG-id when registering etc). > Last time I looked we did not have all nuts and bolts for it, but > let’s give it a try. Yes, reg-id is missing from contact. It would be good to add so that sip proxy can detect if registration is

Re: [SR-Users] Websocket In-dialoge SIP routing failed post network loss due to aliases

2021-06-30 Thread Juha Heinanen
Olle E. Johansson writes: > > Have you checked baresip? > > I don’t recall baresip having a full SIP outbound implementation. baresip is able to register with two outbound proxies and supports gruu (below). What else is needed? -- Juha # TLS 192.168.43.160:49556 -> 192.168.43.160:5061

Re: [SR-Users] Websocket In-dialoge SIP routing failed post network loss due to aliases

2021-06-30 Thread Juha Heinanen
Olle E. Johansson writes: > Full implementation of SIP outbound is the only solution close to > solving this problem in the IETF standards. > However, I have seen no single SIP client that have implemented this, > even though Kamailio supports > it on the server side. The idea is that you

[SR-Users] Websocket In-dialoge SIP routing failed post network loss due to aliases

2021-06-30 Thread Juha Heinanen
Shahid Hussain writes: > Would like to know what is the recommended solution for this problem using > alias or is it a limitation of using alias? Maybe a limitation. Try with SIP User Agents that support gruu and thus identify themselves using sip.instance. -- Juha

Re: [SR-Users] Websocket In-dialoge SIP routing failed post network loss due to aliases

2021-06-30 Thread Shahid Hussain
I am using a JSSIP client and they claim to be implemented RFC-5626. Following are the REGISTER and response messages. Is it possible to confirm the JSSIP client has full implementation of SIP outbound? If it supports fully then I can debug outbound and gruu functionality at Kamailio(I have it

Re: [SR-Users] Websocket In-dialoge SIP routing failed post network loss due to aliases

2021-06-30 Thread Daniel-Constantin Mierla
On 30.06.21 09:28, Juha Heinanen wrote: > Shahid Hussain writes: > >> Would like to know what is the recommended solution for this problem using >> alias or is it a limitation of using alias? > Maybe a limitation. Try with SIP User Agents that support gruu and thus > identify themselves using

Re: [SR-Users] Websocket In-dialoge SIP routing failed post network loss due to aliases

2021-06-30 Thread Olle E. Johansson
> On 30 Jun 2021, at 10:14, Juha Heinanen wrote: > > Olle E. Johansson writes: > >>> Have you checked baresip? >> >> I don’t recall baresip having a full SIP outbound implementation. > > baresip is able to register with two outbound proxies and supports gruu > (below). What else is needed?

Re: [SR-Users] Websocket In-dialoge SIP routing failed post network loss due to aliases

2021-06-30 Thread Olle E. Johansson
> On 30 Jun 2021, at 09:49, Juha Heinanen wrote: > > Olle E. Johansson writes: > >> Full implementation of SIP outbound is the only solution close to >> solving this problem in the IETF standards. >> However, I have seen no single SIP client that have implemented this, >> even though

Re: [SR-Users] Websocket In-dialoge SIP routing failed post network loss due to aliases

2021-06-30 Thread Olle E. Johansson
> On 30 Jun 2021, at 09:10, Shahid Hussain wrote: > > Hi, > Websocket module documentation has a code reference to use aliases for SIP > routing. However, aliases will not work in the following setup and situation. > 1. Kamailio is configured with active and standby node > 2. Ping is

[SR-Users] Websocket In-dialoge SIP routing failed post network loss due to aliases

2021-06-30 Thread Shahid Hussain
Hi, Websocket module documentation has a code reference to use aliases for SIP routing. However, aliases will not work in the following setup and situation. 1. Kamailio is configured with active and standby node 2. Ping is implemented from webclient and kamailio responds with pong. 3. Two clients